Hi Rajan,
Thanks for your reply. I did not enable that option. Once " Send send-receive SDP in mid-call INVITE" is enabled, the first problem (putting caller on hold then resume = no audio) doesnt get resolved.
Also, it does work when its unicast moh and not multicast, therefore telco is allowing a=inactive in the sdp.
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Hi Dennis,
Thanks for your reply. The SIP debug is already attached on my first entry (mmoh-noaudio.txt).
One more thing that i noticed, the moment the call is transferred, i still see the multicast ip 239.1.1.1 on one of the call legs. So its possible the hold session did not break when the call is transferred hence causing the audio to get lost.
***RTP when transfer button is pressed (caller will be put on hold)
VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 121668 121669 17680 6016 216.110.90.193 8.45.210.140 2 121669 121668 17682 16384 10.144.20.6 239.1.1.1
***RTP when the remote party(where the call will be transferred) answered the call
VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 121668 121669 17680 6016 216.110.90.193 8.45.210.140 2 121669 121668 17682 16384 10.144.20.6 239.1.1.1 3 121685 121686 17684 24586 10.144.20.6 10.254.253.12 4 121686 121685 17686 6018 216.110.90.193 8.45.210.140 Found 4 active RTP connections
***RTP when the call has been transferred successfully
VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 121668 121669 17680 6016 216.110.90.193 8.45.210.140 2 121669 121668 17682 17690 10.144.20.6 10.144.20.6 3 121685 121686 17684 17688 10.144.20.6 10.144.20.6 4 121686 121685 17686 6018 216.110.90.193 8.45.210.140 5 121687 121689 17688 17684 10.144.20.6 10.144.20.6 6 121688 121689 17690 17682 10.144.20.6 239.1.1.1 <<<<<<<< caller is no longer on hold yet the multicast ip is still present Found 6 active RTP connections
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Hi Guys,
I hope someone can help me in our issue with our Cube router.
We recently migrated from PRI to SIP. We just learned that MOH from the flash will not work on a SIP trunk in a cube environment. We want to enable multicast moh since unicast might saturate the bandwidth in our datacenter. So we upgraded our 2951 router ios to 15.2 version - c2951-universalk9-mz.SPA.152-4.M10.bin. We were able to make the multicast moh to work, however new issues arised. When we put the call on hold and resume, the audio dissappears. Same thing when we do a supervised transfer (click transfer on the phone > caller will hear moh > dial the number > click transfer),caller will hear nothing as soon as we transfer the call.
I was able to resolve the first issue by creating a new sip profile and enable Early Offer support for voice and video calls (insert MTP if needed) and Require SDP Inactive Exchange for Mid-Call Media Change. So when i put the caller on hold, once i resume, audio is still fine. However, i still cant find any solution on the transfer scenario. I tried enabling other check boxes on the sip profiles but no luck. What puzzles me is when i use unicast moh,the problem gets resolved. So it looks like the issue has something to do with the multicast moh
We are usign CUCM version 8.5 and below is a simple diagram of one of our centers.
ITSP > SIP > Cube >>vpn>> CUCM
We have 100+ sites connected to a single Datacenter where our CUCM is located so as much as possible we dont want to use unicast moh.
I attached the running config of the router and sip debug.
I hope someone can check this one and help me.
Thanks in advance!
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Hi Guys,
Want to ask assistance from you guys again. We migrated from PSTN to SIP last year. Our set-up is like ITSP>>(sip)>>Cube>>>CCMv4>>(sccp)>>Unityv4. All calls inbound and outbound are working. However, when a call from mobile calls our Unity voicemail pilot, it doesnt recognize the digit they pressed. It works if call originates from landline.
DTMF-relay was set to rtp-nte originally on our inbound and outbound dial-peers. I tried to use KPML, Notify etc but nothing worked. i also configured a hardware MTP on our cube and register it to CCM but it still not working. We refered this out to telco and they said that our cube is not confirming rtp-payload 101 if calls originates from TDM. But if it comes from another voip peer, its likely they are sending RFC2833 which requires no rtp-payload 101 to work.
My question is why our cube is not confirming rtp-payload 101? Is there any command in the cube to configure this? Does it have something to do with the version of our CCM and unity(both are version4)? Attached are the running config of the cube and debug results.
I hope someone can help us on this.
Thanks in advance!
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Thanks a lot Les Pruden! This is what im looking for. Our's are VM running on ucs-c. I guess we can now proceed with the project. I appreciate your help sir!
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Hi Mohammed,
Thanks for your reply. Im pretty sure our hardware specs can handle this. What im worried is if the CUCM has limits on the number of SIP trunks that can be added. I know that you can set 15 destinations per trunk, but no mention if there is a maximum number of sip trunks you can set on a cluster.
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Hi Guys!
Good day! I have an upcoming project wherein we will migrate PSTN centres to SIP. These centres are composed of 130 sites and i was advised by my boss that SIP trunk will be installed individually per site. No TEHO, no intersite calling between centres.
My question is, how many sip trunks can be added to CUCM? I cant find any document related to this. We have a single cluster compose of 1 publisher and 9 subscribers. All CUCM version is 8.5.1.
Hope someone can help me with this so i can inform my boss as well.
Thanks!
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Hi EU UC Support, I tried this and it worked! I was puzzled because i checked this before i posted here and im pretty sure the necessary ip were already added. And then i realized that our firewall is Natting the ip of the ccm to a public ip before it reach the voice router. I already amended the needed configurations so everything is working now. Thanks a lot for your help!
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Hi Guys!
Good day! Hope someone can help me with my dilemma involving an ISR router and a very old call manager. We are in the process of migrating several of our centers from PSTN to SIP. Most of these centers are using old Cisco call manager version 4.1, which is not a problem since their cube gateways are mostly 3845 routers. Now, one of the centers have a new Cisco ISR 4351 router which has a uck9 license. I have configured the SIP trunk on the cisco call manager and pointed it to the public ip(reachable from the ccm and phones) of the router. Inbound calls are working however outbound calls do not. I noticed that on my ccsip messages debugs, the ccm keeps on sending invites to the router but it just ignores these messages and sends no response to the callmanager. Now, is it possible that the CCMv4.1 is not compatible with the isr4000 router? If not, am i missing some commands to force the router to reply to the sip invites from the callmanager? Attached are the debugs and running configuration of the router. Please note that the config works on a ccmv4.1 and 3845 voice gateway set-up.
Thanks in advance and i promise to rate all useful tips! =)
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Hi mgogna, Thanks for your reply. I checked the user settings and their transfer type is set to release to switch instead of supervise transfer. Also, voicemail greeting is set to "Take Message" therefor transfer rule will not be applied.
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Hi Guys, Good day! Hope everyone is doing well at the moment. We are having some weird issue with our callmanager and unity connection. They both on version 8.x. Whats happening is that when a call for an extension is forwarded to voicemail (tick VM under Call Forward and Call Pickup Settings), we'll get a 10 second delay in getting response from the cisco unity and instead hear 2x rings. After that, we'll then get the user voicemail greeting. It works fine when we point it to another unity cluster, wherein we get immediate greetings instead of 2 rings. That proves that the issue lies on the cisco unity server but we dont know what settings dictates this kind of behavior. Please see debug isdn q931 sample RTR01Remote# 001252: May 5 11:23:01: ISDN Se0/1/0:23 Q931: RX <- SETUP pd = 8 callref = 0x1741 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x21A3, N/A Plan:ISDN, Type:National Called Party Number i = 0xA1, '4226774443' Plan:ISDN, Type:National 001253: May 5 11:23:01: ISDN Se0/1/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x9741 Channel ID i = 0xA98381 Exclusive, Channel 1 RTR01Remote# 001254: May 5 11:23:01: ISDN Se0/1/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x9741 Progress Ind i = 0x8088 - In-band info or appropriate now available RTR01Remote# 001255: May 5 11:23:11: ISDN Se0/1/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x9741 Display i = 'Connect VM' 001256: May 5 11:23:12: ISDN Se0/1/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x1741 001257: May 5 11:23:12: ISDN Se0/1/0:23 Q931: RX <- STATUS pd = 8 callref = 0x1741 Cause i = 0x80E328 - Information element not implemented Call State i = 0x0A Hope somebody can help us with this. Please advise if additional sample/tests are needed for this. Thanks!
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